The invention relates to a method for the loudness-controlled processing of acoustic signals in acoustic processing equipment, as well as to an apparatus for performing the method according to the preambles of the independent claims. The invention is particularly suitable for use in hearing aids for hearing impaired persons. Entering acoustic signals are processed in such a way that the loudness subjectively received by the hearing impaired person always corresponds to the loudness received by persons with normal hearing.
The idea of loudness-controlled processing of acoustic signals has long been known and has been described by numerous authors, e.g. by N. Dillier et al. in xe2x80x9cJournal of Rehabilitation Research and Developmentxe2x80x9d, vol. 30, No. 1, 1993, pp 100-103. The method is based on the fact persons with normal hearing and with impaired hearing are provided with test signals for evaluating the subjectively received loudness. Harmonic sinusoidal signals or narrow-band noise are used as test signals. The subjectively received loudness is dependent on the signal power and the frequency of a sinusoidal signal, or the frequency of the dominant signal components of a complex signal. The subjective loudness details are determined on a normalized or standard scale with the value range [0, 1]. By comparing the details from a hearing impaired person with those of a reference group of listeners with normal hearing, it is possible to determine hearing impaired-specific, loudness-dependent correcting data. In a matching signal processing method these correcting data are used in order to process for the hearing impaired person the acoustic signals of his environment in the aimed manner. Remarkable intelligibilty improvements were proved in the aforementioned article in the case of intelligibility tests with a group of 13 hearing impaired persons.
Despite the audiological action, the loudness-controlled processing cannot be used in practice in the form known up to now. As described in the aforementioned article, processing takes place by Fourier transformation of short signal segments, the modification of short-time spectra and retransformation of the modified short-time spectra into the time domain. As a result of the segmentwise processing there is a delay of almost 20 ms for the processed signal. This delay is unimportant in intelligibility tests. However, in practice if the hearing impaired person also speaks and perceives his own voice with such a delay, this is completely unacceptable. In the method described in said article the duration of the individual segments is 12.8 ms and it is also possible to drop significantly below this value, because for obtaining a usable short-time spectrum a minimum segment duration of this order of magnitude is vital.
As an alternative to segmentwise processing the starting point was used of subdividing the acoustic signal into subband signals and to process the individual subband signals with separate amplification or gain values. It is known from practical tests that on subdividing into up to three subband signals improvements can be obtained. A subdivision into more subband signals leads to inferior results. A possible reason for this is the discontinuities of the transfer function occurring at the subband boundaries. On comparing the subdivision of the signal into three subband signals with the frequency resolution of short-time spectra of segmentwise processing, it is clear that the potential of the latter cannot be exhausted with the alternative starting point. Even if with the subdivision into more subband signals ways to obtain improved results were found, this would once again lead to the problem of significantly increasing signal delay.
Another aspect for a successful loudness-controlled signal processing is associated with the loudness model used in processing. Unlike simple test signals, the signal power of speech, music and noise is subdivided in time-dependent, complex manner over a wide frequency interval. With a loudness model with said complex signals is associated in time-dependent manner a loudness value, which in the ideal case exactly coincides with the loudness received by listeners with normal hearing. The value determined with the loudness model is used for the time-dependent control of signal processing. The loudness model described in the aforementioned article, apart from the total energy of a signal segment, also takes account of the centre of the short-time spectrum. For calculating the centre of the short-time spectrum use is made of the E. Zwicker bases summarized on pp 153 to 160 of his text book Psychoacoustics, Springer Publishing, Berlin-Heidelbreg-New York, 1990, 1999. From the spectral lines of the short-time spectrum, in a first stage the energies E(z) of the individual frequency groups are formed and then in analogy to the calculation of the centre of gravity in mechanics calculation takes place on the Bark scale z to a centre of the short-time spectrum
c=xcexa3Zxc2x7E(z)/xcexa3E(z)xe2x80x83xe2x80x83(1)
If it was wished to implement this loudness model by subdividing the signal into subband signals, then for processing a band width of 7700 Hz in all it would be necessary to form 21 subband signals of different band width corresponding to the known frequency group width. Besides the aforementioned, sharply rising signal delay, this procedure would require extremely great arithmetical resources. With the presently available technologies for integrated circuits, as for the starting point with segmentwise processing, the transformation into a hearing aid with the existing geometrical dimensions and power consumption is excluded.
The object of the present invention is to provide a method for the loudness-controlled processing of acoustic signals in acoustic processing devices, which can in particular be used in hearing aids. The loudness subjectively received by the hearing aid user should always correspond to the loudness received by a person with normal hearing. In particular the signal delay must be so small that a hearing aid user is not irritated by the delayed perception of his own voice when speaking. There must also be a reduction in the arithmetical resources compared with known methods for the loudness-controlled processing of acoustic signals. In addition, an apparatus for performing the method according to the invention is to be provided.
In the method according to the invention, the processing of the acoustic signal takes place without Fourier transformation, i.e. completely in the time domain and also without subdivision into subband signals. The special nature of the inventive method is that a control quantity x characteristic of the loudness is iteratively calculated and used for controlling a time-dependent correcting filter. The term xe2x80x9citerative calculation procedurexe2x80x9d means that a new value is calculated for each sampling time for the control quantity x using values having the quantities necessary for their calculation in the respectively preceding sampling time. Unlike in the known segmentwise procedure, the loudness-specific control quantity is not only determined as a mean value of successive signal segments, but instead as a continuous time function. The short signal delay of typically 2 ms represents the observation time necessary for a reliable estimated value formation over and beyond the validity time and therefore, unlike in the segmentwise procedure, is not merely the consequence of a disadvantageous characteristic of the selective implementation. The iterative calculation procedure takes place in the inventive method by means of particularly efficient and at the same time original method steps.
The time-dependent correcting filter is controlled in that to the parameters of said filter, new values are allocated at each sampling time by interpolation with the aid of the control quantity x. Unlike in the segmentwise procedure, where the hearing impaired-specific correcting data are stored as amplification values for the individual spectral lines of a short-time spectrum, in the inventive method for well defined values of the control quantity x coefficient sets for prototype filters are predetermined and stored. The transfer functions of these prototype filters pass along the corresponding amplification values, which are determined in the segmentwise method for the individual spectral lines of a short-time spectrum. In the method according to the invention, for characterizing the prototype filters use is made of coefficient sets, whereof it is known that they are suitable for an interpolation, i.e. that the transfer function determined by the interpolated coefficients, in accordance with expectations, passes between the transfer functions, which are determined by the coefficient sets on which the interpolation is based.
Thus, completely new ways are taken by the method according to the invention. The good intelligibility results described in the N. Dillier article are obtained. However, the inventive method also reduces the signal delay to about 2 ms and at the same time drastically reduces the arithmetical resources. It is therefore possible to implement the method according to the invention into a hearing aid of existing construction.
The invention also relates to an apparatus for performing the method according to the invention. This apparatus contains a stage for the iterative calculation of the loudness-characteristic control quantity x and a correcting filter stage controlled in time-dependent manner therewith, which in aimed manner processes incoming acoustic signals. There are various reasons for the aforementioned drastic reduction in the necessary processing resources. Firstly, in the iterative calculation procedure there is no need for the segmentwise buffer storage of the input and output signal. In addition, on storing coefficient sets for the prototype filters, there is also a significant saving compared with the storing of amplification values for the individual spectral lines of the short-time spectra.